[asterisk-bugs] [JIRA] (ASTERISK-22564) directmedia=no in sip.cnf seems to be ignored

[Jon Westgate (JIRA)]

Jon Westgate created ASTERISK-22564:
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             Summary: directmedia=no in sip.cnf seems to be ignored
                 Key: ASTERISK-22564
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22564
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Bridges/bridge_native_rtp
    Affects Versions: 12.0.0-alpha1
         Environment: linux_x64 debian with Asterisk SVN-trunk-r399533
            Reporter: Jon Westgate


I note that this has been reported before and was closed as not a bug.
If this is not a bug, what is the correct way to disable native rtp_bridging on a per sip account basis?
It used to be canreinvite=no.

I can observe on 2 systems running asterisk 12/SVN-trunk-r399533 that with directmedia=no set on both sip trunk and sip client that this happens:

– Executing [07974XXXXXX at wide:1] Dial("SIP/202-00000000", "SIP/JON_34XXXX/07974XXXXXX") in new stack
== Using SIP RTP CoS mark 5
– Called SIP/JON_34XXXX/07974XXXXXX
> 0x7f0bdc018480 – Probation passed - setting RTP source address to [2XXX:XX0:0:X0::5060:2]:23569
– SIP/JON_34XXXX-00000001 is ringing
– SIP/JON_34XXXX-00000001 is making progress passing it to SIP/202-00000000
> 0x7f0bdc018480 – Probation passed - setting RTP source address to [2XXX:XX0:0:X0::5060:2]:23569
> 0x7f0bb003b470 – Probation passed - setting RTP source address to 8X.1X7.X85.X34:11794
– SIP/JON_34XXXX-00000001 answered SIP/202-00000000
– Channel SIP/202-00000000 joined 'simple_bridge' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
– Channel SIP/JON_34XXXX-00000001 joined 'simple_bridge' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
> Bridge 4ff3502f-2411-410e-9f9f-eda20b8b1efb: switching from simple_bridge technology to native_rtp
– Channel SIP/JON_34XXXX-00000001 left 'native_rtp' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
– Channel SIP/202-00000000 left 'native_rtp' basic-bridge <4ff3502f-2411-410e-9f9f-eda20b8b1efb>
== Spawn extension (wide, 07974XXXXXX, 1) exited non-zero on 'SIP/202-00000000'

In this case I have a provider who I connect to via IPV6 and my ip-phone is IPV4-only so a native rtp bridge is never going to work.
however I would like native rtp bridge to work for internal calls.

My outgoing sip trunk:
[JON_34XXXX]
type=peer
remotesecret=password ; Our password to their service
defaultuser=+44145XXXXXXX ; Authentication user for outbound
host=voiceless.aa.net.uk
canreinvite=no
directmedia=no

The phone's config
[202]
secret=xXxXxXxXxXxX
username=202
mailbox=202
nat=force_rport,comedia
type=friend
context=wide
host=dynamic
canreinvite=no
directmedia=no
disallow=all
allow=ulaw
allow=gsm
allow=alaw
dtmfmode=rfc2833

Note that this not only effects asterisk in its IPV6 <> IPV4 bridging modes but also in IPV4 <> IPV4.
I have a scenario where phones exist on a NAT network and the phone server is straddled between an external IP and the internal network.
This causes all kinds of one way audio and dropped calls after random timings.

I guess there may actually be 2 bugs here:
1) Why does directmedia=no have no effect.
2) Why does rtp_bridge think it can bridge IPV4 and IPV6 rtp streams natively.

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